Like you point out its not SIP specific but there are certain aspects of SIP
that we need to take care of and RAQMON does that. let me clarify;
Since it is SIP the session monitoring is media agnostic and need to accomodate
that.
Thats what RAQMON doesin a media gnostioc fashion.
Voice Over IP, Video over IP or Fax over IP and many other
apps fit into the Framework. See it helps you.
Since it is in WG LAst call, would be very happy to see what kind of needs you
have and ensure
that it serves the purpose.
Anwar
-----Original Message-----
From: Romascanu, Dan (Dan)
Sent: Monday, November 01, 2004 2:36 AM
To: ThomasGal(_at_)LumenVox(_dot_)com; Madabhushi Pramod;
sip-implementors-requesto(_at_)cs(_dot_)columbia(_dot_)edu;
sipforum-discussion(_at_)lists(_dot_)su(_dot_)se; ietf(_at_)ietf(_dot_)org
Cc: Siddiqui, Anwar A (Anwar)
Subject: RE: Collecting media statistics for SIP calls?
I agree. This is not a SIP domain specific issue. See my previous answer
pointing to the real-time application QoS monitoring (RAQMON) work in the RMON
MIB WG.
Regards,
Dan
-----Original Message-----
From: ietf-bounces(_at_)ietf(_dot_)org
[mailto:ietf-bounces(_at_)ietf(_dot_)org]On
Behalf Of Thomas Gal
Sent: 31 October, 2004 11:00 PM
To: 'Madabhushi Pramod';
sip-implementors-requesto(_at_)cs(_dot_)columbia(_dot_)edu;
sipforum-discussion(_at_)lists(_dot_)su(_dot_)se; ietf(_at_)ietf(_dot_)org
Subject: RE: Collecting media statistics for SIP calls?
I don't think that's really a SIP domain issue, though
this may have
been adressed somewhere that I'm not aware of.
If you're using RTP to carry the audio than these statistics are
derived from the RTP stream in the context of the sender
(with you being the
receiver) and the sender sends SR (Sender Report) RTCP packets with
interarrival jitter included. Packet count, fraction lost,
and cumulative
number of packets lost are also transmitted in these packets.
Latency would
probably be derived from the NTP timestamp if anything and is
not directly
addressed in RTP to my knowledge. So if you wanted this data
after the fact
the server would have to maintain that information, and it
would probably be
a query at the application level, not really anything to do with SIP.
-Tom
thomasgal(_at_)lumenvox(_dot_)com
-----Original Message-----
From: ietf-bounces(_at_)ietf(_dot_)org
[mailto:ietf-bounces(_at_)ietf(_dot_)org] On
Behalf Of Madabhushi Pramod
Sent: Friday, October 29, 2004 4:09 PM
To: sip-implementors-requesto(_at_)cs(_dot_)columbia(_dot_)edu;
sipforum-discussion(_at_)lists(_dot_)su(_dot_)se; ietf(_at_)ietf(_dot_)org
Subject: Collecting media statistics for SIP calls?
Is there any way by which I call query a SIP endpoint for
media statictics after call termination. I would like to know
details like Jitter, latency, packet loss, packets received,
packets sent etc.
Thanks in advance.
Pramod Madabhushi
ShoreTel communications.
=====
Pramod Madabhushi
email: mpramod(_at_)hotmail(_dot_)com,
madabhushi_p(_at_)yahoo(_dot_)com
Phone:001-408-204-8077
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